This invention relates to telecommunications systems and in particular to an apparatus and method for selecting voice encoding algorithms in such systems.
A recent development in telecommunications technology has been the introduction of asynchronous networks in which traffic is carried in packets or cells, the technique generally being referred to as the asynchronous transfer mode (ATM). In such a network, user traffic is packaged into cells each of which is provided with a header containing supervisory or overhead information. As each packet is filled, it is routed across the network to its desired destination, the routing being determined from the packet header information. A potential problem in such a system is the assembly delay inherent in filling a packet with user information or traffic. For high bit rate users the packets are filled rapidly and the delay is therefore insignificant. For low bit rate users, for example voice traffic which is typically carried in 64 kbit/channels, the assembly delay required to fill a packet can become unacceptably long. One approach to this problem is the insertion of padding or dummy traffic into the packets so as to increase the rate at which they are filled. This is considered undesirable as it is an inefficient use of the available bandwidth. To address this problem various workers have introduced ATM transmission techniques in which information streams, typically voice traffic, from a number of low bit rate users is multiplexed on to a single ATM connection. This increases the rate at which individual ATM packets are filled thus overcoming the assembly delay problem. A particular protocol defining this form of transmission is that described in the ITU-T AAL2 standard recommendation 1.363.2.
In the multiplexed connection, each individual information stream can carry voice information that has been encoded using one of a number of available algorithms. This encoding reduces the bit rate and thus frees up bandwidth to allow higher volume of voice traffic to be accommodated. The range of algorithms available for this purpose enables voice calls to be transported at a corresponding range of bit rates. This offers users the ability to reduce or increase the amount of bandwidth required to support voice calls with a corresponding loss or gain in voice quality. This also allows operators to offer premium voice services at a higher cost to the user. In addition to voice encoding, silence suppression may also be used to reduce the amount of bandwidth required for calls.
It will be appreciated that where a number of users share a single ATM connection for voice traffic, they will not in general all use the same algorithm for voice encoding. Each user will use an algorithm appropriate to the particular call and to the class of service that he or she enjoys. Further, it is possible for more than one algorithm to be employed during an individual call, e.g. to overcome temporary congestion where a call may be xe2x80x98downspeededxe2x80x99 to reduce its coding rate and thus save on bandwidth. As soon as this congestion disappears, the call can be xe2x80x98upspedxe2x80x99 back to the original coding rate. In order to ensure satisfactory operation of such a system, it is of course necessary that the interworking functions (IWF) at both ends of the ATM connection are made aware of the encoding algorithm currently in use on each individual call. This is at present performed from explicit protocol control information carried in the headers of the AAL2 packets transporting the user information streams. The AAL2 Negotiation Procedure (ANP) as defined in the ITU-T standard draft recommendation I.anp is used to define, during the call set up procedure, the meaning of this protocol control information. This meaning is conveyed by reference to a standard profile.
As discussed above, a number of coding algorithms may be used in the course of a single call. To address this scenario, the ITU-T recommendation currently designated as I.trunk defines a number of coding profiles each containing a number of coding algorithms that belong to a particular coding family. A predetermined profile is selected for a call and conveyed from the transmitting IWF to the receiving IWF by means of the ANP. Any algorithm within that profile is then available for use during that call. Each of the algorithms within a profile can be identified within the header of an AAL2 CPS packet (or minicell) by reference to two fields. These are the Length Indicator (LI) field and the User-to-User Information (UUI) field whose meanings are fixed by the ANP such that each algorithm can be uniquely identified from CPS packet to CPS packet throughout a call. However, it is a prerequisite of this procedure that only the standard algorithm profiles can be employed.
While this approach provides an effective method of identifying coding algorithms to the receiving interworking function (IWF), it suffers from the disadvantage that only a small number of encoding algorithm profiles can be identified in this way and there is no mechanism for identifying a non-standard profile of encoding algorithms. A voice call may thus have to employ a less than ideal profile as this is all that is currently available from the standard profile set.
An object of the invention is to minimise or to overcome the above disadvantage.
A further object of the invention is to provide an improved method of identifying algorithms used for voice encoding in a telecommunications system.
According to the invention there is provided a method of transmitting a plurality of narrow band digital traffic channels in packets over an asynchronous packet switched network between a transmitter station and a receiver station, wherein each said channel is encoded via one of a set of encoding algorithms associated with that channel, wherein the identity of the algorithnms in each channel set of algorithms is conveyed to the receiver at call set-up between the transmitter and receiver, and wherein the identity of that algorithm emploed to encode a said channel is conveyed via header information provided in the packets carrying that channel.
According to another aspect of the invention there is provided a method of transmitting a plurality of narrow band digital traffic channels over an asynchronous packet switched network between a transmitter station and a receiver station, the method comprising;
at the transmitter station, selecting a group or profile of encoding algorithms for each said channel from a set of encoding algorithms;
encoding each said channel with an algorithm selected from its corresponding group of algorithms;
multiplexing the encoded channels into a common connection over the asynchronous network;
providing a service specific convergence sublayer (SSCS) selection parameter field incorporating for each said channel a plurality of encoding algorithm identifier fields each indicative of a respective algorithm in the group selected for that channel and carrying further information whereby the receiver station may uniquely identify the algorithm currently in use on that channel;
at a receiving station demultiplexing the encoded channels;
determining from the information in the SSCS parameter field the identity of the encoding algorithm currently employed for that channel; and decoding and recovering each said channel via its respective algorithm.
The service specific convergence sublayer selection parameter field is conveyed via an AAL2 negotiating procedure (ANP) established between the transmitter and receiver for allocating user channels therebetween.
According to another aspect of the invention there is provided a method of transmitting a narrow band digital traffic channel between a transmitter station and a receiver station over an asynchronous broad band network, the method including:
at the transmitting station, selecting a group or profile of encoding algorithms for said channel from a set of encoding algorithms and encoding said channel with an algorithm selected from its corresponding group of algorithms;
performing a negotiation with the receiver station to effect allocation of a transmission channel therebetween;
providing a service specific convergence sublayer (SSCS) selection parameter field incorporating for said channel a plurality of encoding algorithm identifier fields each indicative of an algorithm for use by that channel and conveying that field via the negotiating procedure to the receiver;
packetising the encoded channel into packets and providing each said packet with a header containing information indicative of the algorithm used to encode the channel;
transmitting the packets over the allocated transmission channel to the receiver station;
at the receiver station, storing the service specific convergence sublayer (SSCS) selection parameter field information;
determining from each received packet header and from the stored information the identity of the algorithm employed to encode the channel; and
decoding that channel via that algorithm so as to recover the channel.
According to another aspect of the invention there is provided a method of transmitting a narrow band digital traffic channel over an asynchronous packet switched network from a transmitter station to a receiver station, the method comprising;
at the transmitter station selecting a group or profile of encoding algorithms for use on a said channel;
encoding the channel via a said algorithm selected from the group;
packetising the encoded channel into packets or cells;
providing each said packet with a header incorporating information means identifying that algorithm selected from the group and currently in use to encode the channel, according to the information in the SSCS parameter field;
transmitting each said packet from the transmitter station to the receiver station;
at the receiver station, determining from the information received in the SSCS parameter field and the information provided in the header of each said packet the algorithm used to encode the channel; and
decoding and recovering the channel via that algorithm.
According to a further aspect of the invention there is provided apparatus for transmitting a plurality of narrow band digital traffic channels over an asynchronous packet switched network, the apparatus comprising;
a transmitting station, having means for selecting a group or profile of encoding algorithms for each said channel from a set of encoding algorithms;
means for encoding each said channel with an algorithm selected from its corresponding group of algorithms;
means for multiplexing the encoded channels into a common connection over the asynchronous network;
means for providing a negotiation procedure between the transmitter and receiver to convey, at the time of setting up individual calls, a service specific convergence sublayer (SSCS) selection parameter field incorporating for each said channel a plurality of encoding algorithm identifier fields each indicative of a respective algorithm in the group that may be selected for that channel at any time during the call;
and a means for providing a unique identification of the algorithm currently in use for a call via information carried in a packet length indicator field and a user to user information (UUI) field within the headers of packets;
and a receiver station, having means for demultiplexing the encoded channels;
means for determining from the information carried in the packet length indicator field and the user to user information field contained in the headers of the packets associated with each said channel the encoding algorithm currently employed for that channel; and means for decoding and recovering each said channel via its respective algorithm.
Typically, the narrow band channels comprise 64 kbit/s or sub-64 kbit/s voice channels.
Identification of an algorithm at the receiver station may be performed implicitly from the packet length indicator, explicitly from a UUI field or from a combination of these techniques.
The technique provides an effective method of encoding and decoding channels using non-standard or custom profiles and of conveying the identity of the encoding algorithm currently in use from the transmitter to the receiver. Furthermore, the technique is fully compatible with existing asynchronous transfer mode standards.